• DocumentCode
    1867349
  • Title

    An analog VLSI architecture for auditory based feature extraction

  • Author

    Kumar, Nagendra ; Himmelbauer, Wolfgang ; Cauwenberghs, Gert ; Andreou, Andreas G.

  • Author_Institution
    Center for Language & Speech Process., Johns Hopkins Univ., Baltimore, MD, USA
  • Volume
    5
  • fYear
    1997
  • fDate
    21-24 Apr 1997
  • Firstpage
    4081
  • Abstract
    We have developed a low power analog VLSI chip for real time signal processing motivated by the principles of the human auditory system. An analog cochlear filter bank (which is implemented on the chip) decomposes the input audio signal into several frequency bands that have almost equal bandwidth on a log scale. This step is thus similar to computing the wavelet transform. The chip then computes signal energies and zero crossing time intervals of frequency components in a cochlear filter bank. The chip is intended to work as a front-end of a speech recognition system. We include experimental results on a VLSI implementation of the auditory front-end. We present speech recognition results on the TI-DIGITS database obtained from computer simulations which model the functionality of the feature extraction VLSI hardware. We use hidden Markov models (HMM) in combination with linear discriminant analysis (LDA) for the recognizer design
  • Keywords
    MOS integrated circuits; VLSI; analogue integrated circuits; band-pass filters; ear; feature extraction; hearing; hidden Markov models; speech processing; speech recognition; MOSIS; TI-DIGITS database; analog VLSI architecture; analog cochlear filter bank; auditory based feature extraction; auditory front-end; bandwidth; computer simulations; experimental results; frequency bands; frequency components; hidden Markov models; human auditory system; input audio signal; linear discriminant analysis; low power analog VLSI chip; real time signal processing; signal energies; speech recognition system; wavelet transform; zero crossing time intervals; Feature extraction; Filter bank; Frequency; Hidden Markov models; Humans; Linear discriminant analysis; Real time systems; Signal processing; Speech recognition; Very large scale integration;
  • fLanguage
    English
  • Publisher
    ieee
  • Conference_Titel
    Acoustics, Speech, and Signal Processing, 1997. ICASSP-97., 1997 IEEE International Conference on
  • Conference_Location
    Munich
  • ISSN
    1520-6149
  • Print_ISBN
    0-8186-7919-0
  • Type

    conf

  • DOI
    10.1109/ICASSP.1997.604843
  • Filename
    604843